## java fft problem

Started by in Audio Signal Processing8 years ago 3 replies

I am trying to implement some algorithm in java that is originally done in matlab. One of the parts is fast fourier transform. In matlab code it...

I am trying to implement some algorithm in java that is originally done in matlab. One of the parts is fast fourier transform. In matlab code it looks as following: FY = fft(X,44100); Where according to http://www.mathworks.com/help/techdoc/ref/fft.html : Y = fft(X,n) returns the n-point DFT. fft(X) is equivalent to fft(X, n) where n is the size of X in the first nonsingleton dimension. If the l...

## Extracting SNR from FFT

Started by in Audio Signal Processing12 years ago 2 replies

Dear All Does anyone here already coded a SNR computation from FFT bins ? I already coded a FFT algo in C, it is ok, but I cannot succeed in...

Dear All Does anyone here already coded a SNR computation from FFT bins ? I already coded a FFT algo in C, it is ok, but I cannot succeed in coding a SNR extraction from the FFT bins. I use a blackman harris 7 windows (7 bins in signal) My FFT table fft[] represents all fft bins in dB - I search the max bin value from FFT table fft[] - for(-3 bins to +3 bins around the above max bin)...

## Decent FFT code

Started by in Audio Signal Processing9 years ago

I'm building a guitar toy and I need an FFT for a dsPIC33 series processor and the Microchip C30 library FFT does not work. (Seriously, whoda...

I'm building a guitar toy and I need an FFT for a dsPIC33 series processor and the Microchip C30 library FFT does not work. (Seriously, whoda thunk?) Being on the lazy side wrt coding (hardware engineer) I was hoping for a plug and play fft module. Anyone know of a good C FFT program that might port to MPLAB C# enviroment? -Cuetek

## [general] arm fft doubt

Started by in Audio Signal Processing12 years ago

dear all, we need to use fft algo. for arm9e processor. we got it from: http://www.lartmaker.nl/projects/fft-arm/ but we aren't able to...

dear all, we need to use fft algo. for arm9e processor. we got it from: http://www.lartmaker.nl/projects/fft-arm/ but we aren't able to build it in windows (codewarrior), as some of the header files are linux specific. is there any code available for windows? or can anybody help in building the same? also, is this code faster than the fft library provided with arm rvds? any help wo...

## FFT Frequency Range

Started by in Audio Signal Processing8 years ago 1 reply

Hi, I have one very simple question. I hope some one will clearify my doubts. The range of frequencies covered in the output record from the...

Hi, I have one very simple question. I hope some one will clearify my doubts. The range of frequencies covered in the output record from the FFT is 0 to 1/2 the sample rate of the acquired data record. For example, a sample rate of 20 MS/s (megasamples per second) would give an FFT range of 0 to 10 MHz. Does it mean I can only see the proper FFT peak of the signal with maximum frequenc...

## Re: Block Floating point FFT or 32 bits precision FFT

Started by in Audio Signal Processing17 years ago

> Date: Wed, 5 Dec 2001 11:47:49 +0100 > From: "Curl" > Subject: Block Floating point FFT or 32 bits precision FFT ? > > Sorry, this is my...

> Date: Wed, 5 Dec 2001 11:47:49 +0100 > From: "Curl" > Subject: Block Floating point FFT or 32 bits precision FFT ? > > Sorry, this is my second question in two days .. > I'd like to know if someone tried this two algorithms : Block Floating > point FFT and 32 bits (double precision) FFT.. > Which gives the better result ? Speak

## 16 bit Fixed point FFT

Started by in Audio Signal Processing13 years ago

Hello, I have a query regarding 1024 point FFT implementation with 16 bits input data( real and imagainary) and the twiddle factors are also...

Hello, I have a query regarding 1024 point FFT implementation with 16 bits input data( real and imagainary) and the twiddle factors are also 16 bits(real and imaginary). The algorithm i am using is Decimation in Frequency Radix-4 FFT. Hence the no. of stages in the FFT is 5. Please let me know how the scaling or rounding of data should be done to fit the final result also in 16 bits. T

## FFT on a narrow band

Started by in Audio Signal Processing16 years ago 3 replies

Does anybody know if it is possible to calculate a FFT on a certain bandwith, less that the sampling frequency? For example: I only have the...

Does anybody know if it is possible to calculate a FFT on a certain bandwith, less that the sampling frequency? For example: I only have the resources to do a 64 point FFT, but I know the energy I want to measure is between 1kHz and 2kHz (fs = 8 kHz). Can I do a 64 point FFT on the band 1kHz ? 2kHz ? Rgds JEAN

## zeroing the FFT

Started by in Audio Signal Processing18 years ago 32 replies

I have an FFT algorithm that works for example only for 1024, but I have only , say 900 discrete values to calculate the FFT on. I put the...

I have an FFT algorithm that works for example only for 1024, but I have only , say 900 discrete values to calculate the FFT on. I put the other 124 values to zero (from position 901 to 1024). The results are different from the FFT done on the 900 samples (1024 frequency samples). Does anybody know how to get it done? Thanks, Radu

## Hann windowed FFT from 'rectangular' one ?

Started by in Audio Signal Processing13 years ago 6 replies

Hello, This is my first post in this group so let me say hi to everybody :) ... and of course I have a question, probably simple but...

Hello, This is my first post in this group so let me say hi to everybody :) ... and of course I have a question, probably simple but ... (how) can I get hann-windowed complex FFT values from the plain (unwindowed) complex FFT ? In other words, having complex spectra X = fft(x, N), can I get hann-windowed complex spectra Xh from X ? Thanks in advance, Best regards, PrzeM

## FFT and EQ

Started by in Audio Signal Processing14 years ago 5 replies

Hello, I have a short question for you. I am working with blocks of audiosamples which will be calculated like this. Audiosamples-in ->...

Hello, I have a short question for you. I am working with blocks of audiosamples which will be calculated like this. Audiosamples-in -> windowing -> FFT -> iFFT -> windowing -> Audiosamples-out. This works fine. I.e. the result of audiosamples-out after all is like the starting point audiosamples-in. Now, I want to build up a FFT EQ. I thought that I just simple set the cal

## Block Floating point FFT or 32 bits precision FFT ?

Started by in Audio Signal Processing17 years ago 1 reply

Hello ! Sorry, this is my second question in two days .. I'd like to know if someone tried this two algorithms : Block Floating point FFT and...

Hello ! Sorry, this is my second question in two days .. I'd like to know if someone tried this two algorithms : Block Floating point FFT and 32 bits (double precision) FFT.. Which gives the better result ? Thank you for your advice..

## FFT precision

Started by in Audio Signal Processing17 years ago 4 replies

Hello My question may be stupid.. I have overflow problem with 16bits IFFT (inverse transform) In audio applications : Is FFT precision at...

Hello My question may be stupid.. I have overflow problem with 16bits IFFT (inverse transform) In audio applications : Is FFT precision at least 32 bits ?? Or (in other words) can you do audio processing with 16 bits FFT ?? Thank you

## a MPEG2 question

Started by in Audio Signal Processing18 years ago

Dear all, I am a newcomer in DSP, and now working on MPEG-2 Audio Encoder. In ISO11172-3 encoding algorithms(Layer II and Layer III), before...

Dear all, I am a newcomer in DSP, and now working on MPEG-2 Audio Encoder. In ISO11172-3 encoding algorithms(Layer II and Layer III), before FFT transfer, there are 1152 samples collected, but only 1024 points FFT are employed, what shall we do with the remaining 128 samples, just ingnored or shall overlapped by Module Reduction N? Before FFT an Hanning window is firstly applied. Best R

## FFT's

Started by in Audio Signal Processing14 years ago 4 replies

Hey all: I'm a newbie here, and have a FFT related question. Some background first. I've written a VB program that takes an FFT of a time...

Hey all: I'm a newbie here, and have a FFT related question. Some background first. I've written a VB program that takes an FFT of a time function comprised of fixed pulse width impulses. The timing of the impulses vary from pulse to pulse in a predetermined arrangement, and there are only two type of impulses, call them Pulse A and Pulse B Now if I arrange these pulses like this:

## how to comput 2N point array by using N point fft?

Started by in Audio Signal Processing13 years ago

everyone: hi. I want to save time in the process of fft,I know there is one method of using a complex N point fft to comput a real 2N...

everyone: hi. I want to save time in the process of fft,I know there is one method of using a complex N point fft to comput a real 2N point array.But I dont know how to do it .can anyone help me?

## coherent/incoherent gain of fft window

Started by in Audio Signal Processing16 years ago

Hi I'm right now dealing wiht calculation of signal to noise ration from an fft transform. Specificly I would like to know more about...

Hi I'm right now dealing wiht calculation of signal to noise ration from an fft transform. Specificly I would like to know more about the coherent and incoherent gain of a fft window (such as Hanning etc.). I have read that if you are calculating the SNR by dividing the signal bin with the sum of the power of the other bins, then you also have to include a correction factor due to the dif

## STFT or FFT Waterfall for loudspeaker?

Started by in Audio Signal Processing9 years ago 2 replies

Dears, I wonder to know in those audio test equipment (e.g. CLIO, Audio Precision), they have a function called Waterfall or CSD. It is for...

Dears, I wonder to know in those audio test equipment (e.g. CLIO, Audio Precision), they have a function called Waterfall or CSD. It is for analyzing a speaker time-frequency relation from an impulse. Do they use FFT or STFT to generate the spectrum? Also, what windows do they use? I tried to FFT an impulse with shifting square window but I got a lot of low frequency noise after the first F...

## FFT output interpretation (FFTW v. Ooura)

Started by in Audio Signal Processing11 years ago 14 replies

I'm new(ish) to FFT and to broaden my knowledge/experience I've been comparing FFTW and the Ooura FFT library. I'm getting a result with the...

I'm new(ish) to FFT and to broaden my knowledge/experience I've been comparing FFTW and the Ooura FFT library. I'm getting a result with the latter that I don't quite understand, however. I have been analysing 1024 samples from a WAV file (16-bit, sr 44100) using FFTW's real-to-complex transform and the equivalent rdft from the Ooura library. I get the same output apart from the DC bin. FFTW gives...

## Frequency response using FFT

Started by in Audio Signal Processing12 years ago 1 reply

I try to get a frequency response using FFT(output)/FFT(input). My input signal is white noise. This works pretty good in low frequency. However...

I try to get a frequency response using FFT(output)/FFT(input). My input signal is white noise. This works pretty good in low frequency. However in high frequency (next to the sampling frequency) the gain curve goes fast to infinite but it shouldn't. Moreover the phase curve is not exactly correct. It has the right form but it goes about 200 degrees too high. How should improve the results? Tha...